Revert of Adding debug dump to audio network adaptor.

webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

Index: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
deleted file mode 100644
index 9992e2dbd7dc93cb0872709541af81481368a9d8..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
-
-#include "webrtc/base/checks.h"
-
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
-#else
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
-#endif
-#endif
-
-namespace webrtc {
-
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-namespace {
-
-using audio_network_adaptor::debug_dump::Event;
-using audio_network_adaptor::debug_dump::NetworkMetrics;
-using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
-
-void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
-  RTC_CHECK(dump_file->is_open());
-  std::string dump_data;
-  event.SerializeToString(&dump_data);
-  int32_t size = event.ByteSize();
-  dump_file->Write(&size, sizeof(size));
-  dump_file->Write(dump_data.data(), dump_data.length());
-}
-
-}  // namespace
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-
-class DebugDumpWriterImpl final : public DebugDumpWriter {
- public:
-  explicit DebugDumpWriterImpl(FILE* file_handle);
-  ~DebugDumpWriterImpl() override = default;
-
-  void DumpEncoderRuntimeConfig(
-      const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
-      int64_t timestamp) override;
-
-  void DumpNetworkMetrics(const Controller::NetworkMetrics& metrics,
-                          int64_t timestamp) override;
-
- private:
-  std::unique_ptr<FileWrapper> dump_file_;
-};
-
-DebugDumpWriterImpl::DebugDumpWriterImpl(FILE* file_handle)
-    : dump_file_(FileWrapper::Create()) {
-#ifndef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-  RTC_DCHECK(false);
-#endif
-  dump_file_->OpenFromFileHandle(file_handle);
-  RTC_CHECK(dump_file_->is_open());
-}
-
-void DebugDumpWriterImpl::DumpNetworkMetrics(
-    const Controller::NetworkMetrics& metrics,
-    int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-  Event event;
-  event.set_timestamp(timestamp);
-  event.set_type(Event::NETWORK_METRICS);
-  auto dump_metrics = event.mutable_network_metrics();
-
-  if (metrics.uplink_bandwidth_bps)
-    dump_metrics->set_uplink_bandwidth_bps(*metrics.uplink_bandwidth_bps);
-
-  if (metrics.uplink_packet_loss_fraction) {
-    dump_metrics->set_uplink_packet_loss_fraction(
-        *metrics.uplink_packet_loss_fraction);
-  }
-
-  if (metrics.target_audio_bitrate_bps) {
-    dump_metrics->set_target_audio_bitrate_bps(
-        *metrics.target_audio_bitrate_bps);
-  }
-
-  if (metrics.rtt_ms)
-    dump_metrics->set_rtt_ms(*metrics.rtt_ms);
-
-  DumpEventToFile(event, dump_file_.get());
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-}
-
-void DebugDumpWriterImpl::DumpEncoderRuntimeConfig(
-    const AudioNetworkAdaptor::EncoderRuntimeConfig& config,
-    int64_t timestamp) {
-#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-  Event event;
-  event.set_timestamp(timestamp);
-  event.set_type(Event::ENCODER_RUNTIME_CONFIG);
-  auto dump_config = event.mutable_encoder_runtime_config();
-
-  if (config.bitrate_bps)
-    dump_config->set_bitrate_bps(*config.bitrate_bps);
-
-  if (config.frame_length_ms)
-    dump_config->set_frame_length_ms(*config.frame_length_ms);
-
-  if (config.uplink_packet_loss_fraction) {
-    dump_config->set_uplink_packet_loss_fraction(
-        *config.uplink_packet_loss_fraction);
-  }
-
-  if (config.enable_fec)
-    dump_config->set_enable_fec(*config.enable_fec);
-
-  if (config.enable_dtx)
-    dump_config->set_enable_dtx(*config.enable_dtx);
-
-  if (config.num_channels)
-    dump_config->set_num_channels(*config.num_channels);
-
-  DumpEventToFile(event, dump_file_.get());
-#endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
-}
-
-std::unique_ptr<DebugDumpWriter> DebugDumpWriter::Create(FILE* file_handle) {
-  return std::unique_ptr<DebugDumpWriter>(new DebugDumpWriterImpl(file_handle));
-}
-
-}  // namespace webrtc